move the APM to chrome.

content/renderer/media/webrtc_audio_processor.cc

Index: content/renderer/media/webrtc_audio_processor.cc
diff --git a/content/renderer/media/webrtc_audio_processor.cc b/content/renderer/media/webrtc_audio_processor.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b1456d3bc4aabfc078f0b5e29eaabbf39df9070c
--- /dev/null
+++ b/content/renderer/media/webrtc_audio_processor.cc
@@ -0,0 +1,424 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "content/renderer/media/webrtc_audio_processor.h"
+
+#include "base/command_line.h"
+#include "base/debug/trace_event.h"
+#include "content/public/common/content_switches.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_converter.h"
+#include "media/base/audio_fifo.h"
+#include "media/base/channel_layout.h"
+
+namespace content {
+
+namespace {
+
+using webrtc::AudioProcessing;
+using webrtc::MediaConstraintsInterface;
+
+#if defined(ANDROID)
+const int kAudioProcessingSampleRate = 16000;
+#else
+const int kAudioProcessingSampleRate = 32000;
+#endif
+const int kAudioProcessingNumberOfChannel = 1;
+
+const int kMaxNumberOfBuffersInFifo = 2;
+
+bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints,
+                                const std::string& key) {
+  bool value = false;
+  return webrtc::FindConstraint(constraints, key, &value, NULL) && value;
+}
+
+// Extract all this methods to a helper class.
+void EnableEchoCancellation(AudioProcessing* audio_processing) {
+  DCHECK(audio_processing);
+#if defined(IOS) || defined(ANDROID)
+  // Mobile devices are using AECM.
+  if (audio_processing->echo_control_mobile()->Enable(true))
+    NOTREACHED();
+
+  if (audio_processing->echo_control_mobile()->set_routing_mode(
+      webrtc::EchoControlMobile::kSpeakerphone))
+    NOTREACHED();
+
+  return;
+#endif
+  if (audio_processing->echo_cancellation()->Enable(true))
+    NOTREACHED();
+  if (audio_processing->echo_cancellation()->set_suppression_level(
+      webrtc::EchoCancellation::kHighSuppression))
+    NOTREACHED();
+
+  // Enable the metrics for AEC.
+  if (audio_processing->echo_cancellation()->enable_metrics(true))
+    NOTREACHED();
+  if (audio_processing->echo_cancellation()->enable_delay_logging(true))
+    NOTREACHED();
+}
+
+void EnableNoiseSuppression(AudioProcessing* audio_processing) {
+  DCHECK(audio_processing);
+  if (audio_processing->noise_suppression()->set_level(
+      webrtc::NoiseSuppression::kHigh))
+    NOTREACHED();
+
+  if (audio_processing->noise_suppression()->Enable(true))
+    NOTREACHED();
+}
+
+void EnableHighPassFilter(AudioProcessing* audio_processing) {
+  DCHECK(audio_processing);
+  if (audio_processing->high_pass_filter()->Enable(true))
+     NOTREACHED();
+}
+
+// TODO(xians): stereo swapping
+void EnableTypingDetection(AudioProcessing* audio_processing) {
+  DCHECK(audio_processing);
+  if (audio_processing->voice_detection()->Enable(true))
+    NOTREACHED();
+
+  if (audio_processing->voice_detection()->set_likelihood(
+      webrtc::VoiceDetection::kVeryLowLikelihood))
+    NOTREACHED();
+}
+
+void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) {
+  DCHECK(audio_processing);
+  webrtc::Config config;
+  config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
+  audio_processing->SetExtraOptions(config);
+}
+
+void StartAecDump(AudioProcessing* audio_processin) {
+  static const char kAecDumpFilename[] = "/tmp/audio.aecdump";

[Henrik Grunell, 2013/10/31 11:56:12]

This should be different for different platforms. See webrtcvoiceengine.cc in
libjingle ~line 138. Also move the comment and todo (for crbug.com/264223) here.

+  if (audio_processin->StartDebugRecording(kAecDumpFilename))
+    LOG(ERROR) << "Fail to start AEC debug recording";
+}
+
+void StopAecDump(AudioProcessing* audio_processin) {
+  if (audio_processin->StopDebugRecording())
+    LOG(ERROR) << "Fail to stop AEC debug recording";
+}
+
+}  // namespace
+
+class WebRtcAudioProcessor::WebRtcAudioConverter
+    : public media::AudioConverter::InputCallback {
+ public:
+  WebRtcAudioConverter(const media::AudioParameters& source_params,
+                       const media::AudioParameters& sink_params) {
+    source_params_ = source_params;
+    sink_params_ = sink_params;
+
+    // Create the audio converter which is responsible for down-mixing and
+    // resampling.
+    audio_converter_.reset(
+        new media::AudioConverter(source_params, sink_params_, false));
+    audio_converter_->AddInput(this);
+
+    // Create and initialize audio fifo and audio bus wrapper.
+    // The size of the FIFO should be at least twice of the source buffer size
+    // or twice of the sink buffer size.
+    int buffer_size = std::max(
+        kMaxNumberOfBuffersInFifo * source_params.frames_per_buffer(),
+        kMaxNumberOfBuffersInFifo * sink_params_.frames_per_buffer());
+    fifo_.reset(new media::AudioFifo(source_params.channels(), buffer_size));
+    // TODO(xians): Use CreateWrapper to save one memcpy.
+    audio_wrapper_ = media::AudioBus::Create(sink_params_.channels(),
+                                             sink_params_.frames_per_buffer());
+  }
+
+  ~WebRtcAudioConverter() {
+    audio_converter_->RemoveInput(this);
+  }
+
+  void Push(media::AudioBus* audio_source) {
+    DCHECK(fifo_->frames() + audio_source->frames() <= fifo_->max_frames());
+    fifo_->Push(audio_source);
+  }
+
+  bool Convert() {
+    // Return false if there is no 10ms data in the FIFO.
+    if (fifo_->frames() < (source_params_.sample_rate() / 100))
+      return false;
+
+    // Convert 10ms data to the output format, this will trigger ProvideInput().
+    audio_converter_->Convert(audio_wrapper_.get());
+
+    // TODO(xians): A better way to handle the interleaved and deinterleaved
+    // format switching.
+    audio_wrapper_->ToInterleaved(audio_wrapper_->frames(), 2,
+                                  audio_frame_.data_);
+
+    audio_frame_.samples_per_channel_ = sink_params_.frames_per_buffer();
+    audio_frame_.sample_rate_hz_ = sink_params_.sample_rate();
+    audio_frame_.speech_type_ = webrtc::AudioFrame::kNormalSpeech;
+    audio_frame_.vad_activity_ = webrtc::AudioFrame::kVadUnknown;
+    audio_frame_.num_channels_ = sink_params_.channels();
+//    audio_frame_.interleaved_ = false;
+
+    return true;
+  }
+
+  webrtc::AudioFrame* audio_frame() { return &audio_frame_; }
+  const media::AudioParameters& source_parameters() const {
+    return source_params_;
+  }
+  const media::AudioParameters& sink_parameters() const {
+    return sink_params_;
+  }
+
+ private:
+  // AudioConverter::InputCallback implementation.
+  virtual double ProvideInput(media::AudioBus* audio_bus,
+                              base::TimeDelta buffer_delay) {
+    // The first Convert() can trigger ProvideInput two times, use SincResampler
+    // to fix the problem.
+    if (fifo_->frames() < audio_bus->frames())
+      return 0;
+
+    fifo_->Consume(audio_bus, 0, audio_bus->frames());
+    return 1.0;
+  }
+
+  webrtc::AudioFrame audio_frame_;
+
+  // TODO(xians): consider using SincResampler to save some memcpy.
+  // Handles mixing and resampling between input and output parameters.
+  scoped_ptr<media::AudioConverter> audio_converter_;
+  scoped_ptr<media::AudioBus> audio_wrapper_;
+  scoped_ptr<media::AudioFifo> fifo_;
+
+  media::AudioParameters source_params_;
+  media::AudioParameters sink_params_;
+};
+
+WebRtcAudioProcessor::WebRtcAudioProcessor(
+    const webrtc::MediaConstraintsInterface* constraints)
+    : render_delay_ms_(0) {
+  InitializeAudioProcessingModule(constraints);
+}
+
+WebRtcAudioProcessor::~WebRtcAudioProcessor() {
+  StopAudioProcessing();
+}
+
+void WebRtcAudioProcessor::SetFormat(
+    const media::AudioParameters& source_params) {
+  DCHECK(source_params.IsValid());
+
+  // Create and initialize audio converter.
+  int sink_sample_rate = audio_processing_.get() ?
+      kAudioProcessingSampleRate : source_params.sample_rate();
+  media::ChannelLayout sink_channel_layout = audio_processing_.get() ?
+      media::CHANNEL_LAYOUT_MONO : source_params.channel_layout();
+
+  // WebRtc is using 10ms data as its native packet size.
+  media::AudioParameters sink_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY, sink_channel_layout,
+      sink_sample_rate, 16, sink_sample_rate / 100);
+  capture_converter_.reset(
+      new WebRtcAudioConverter(source_params, sink_params));
+}
+
+void WebRtcAudioProcessor::Push(media::AudioBus* audio_source) {
+  DCHECK(capture_converter_.get());
+  capture_converter_->Push(audio_source);
+}
+
+bool WebRtcAudioProcessor::ProcessAndConsume10MsData(
+    int capture_audio_delay_ms, int volume, bool key_pressed) {
+  TRACE_EVENT0("audio",
+               "WebRtcAudioProcessor::ProcessAndConsume10MsData");
+
+  if (!capture_converter_->Convert())
+    return false;
+
+  Process10MsData(capture_audio_delay_ms, volume, key_pressed);
+
+  return true;
+}
+
+const int16* WebRtcAudioProcessor::OutputBuffer() const {
+  return &capture_converter_->audio_frame()->data_[0];
+}
+
+const media::AudioParameters&
+WebRtcAudioProcessor::OutputFormat() const {
+  return capture_converter_->sink_parameters();
+}
+
+
+void WebRtcAudioProcessor::Process10MsData(int capture_audio_delay_ms,
+                                                   int volume,
+                                                   bool key_pressed) {
+  if (!audio_processing_.get())
+    return;
+
+  // TODO(xians): Add a DCHECK it is 10ms data chunk.
+
+  TRACE_EVENT0("audio", "WebRtcAudioProcessor::Process10MsData");
+  DCHECK_EQ(audio_processing_->sample_rate_hz(),
+            capture_converter_->sink_parameters().sample_rate());
+  DCHECK_EQ(audio_processing_->num_input_channels(),
+            capture_converter_->sink_parameters().channels());
+  DCHECK_EQ(audio_processing_->num_output_channels(),
+            capture_converter_->sink_parameters().channels());
+
+  // TODO(xians): Sum the capture delay and render delay.
+  int total_delay_ms = 0;
+  {
+    base::AutoLock auto_lock(lock_);
+    total_delay_ms = capture_audio_delay_ms + render_delay_ms_;
+  }
+
+  audio_processing_->set_stream_delay_ms(total_delay_ms);
+  webrtc::GainControl* agc = audio_processing_->gain_control();
+  if (agc->set_stream_analog_level(volume))
+    NOTREACHED();
+  int err = audio_processing_->ProcessStream(
+      capture_converter_->audio_frame());
+  if (err) {
+    NOTREACHED() << "ProcessStream() error: " << err;
+  }
+
+  // TODO(xians): Get the new volume and pass it to the capturer.
+//  new_volume_ = agc->stream_analog_level();
+
+  // TODO(xians): Handle the typing detection event here.
+  // TypingDetection(key_pressed);
+}
+
+void WebRtcAudioProcessor::FeedRenderDataToAudioProcessing(
+    const int16* render_audio, int sample_rate, int number_of_channels,
+    int number_of_frames, int render_delay_ms) {
+  if (!audio_processing_.get())
+    return;
+
+  TRACE_EVENT0("audio",
+               "WebRtcAudioProcessor::FeedRenderDataToAudioProcessing");
+  {
+    base::AutoLock auto_lock(lock_);
+    render_delay_ms_ = render_delay_ms;
+  }
+
+  InitializeRenderConverterIfNeeded(sample_rate, number_of_channels,
+                                    number_of_frames);
+  DCHECK(render_converter_.get());
+
+  // FIXME. This is crazy, a few extra copy and interleave/deinterleave.
+  scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create(
+      number_of_channels, number_of_frames);
+  data_bus->FromInterleaved(render_audio,
+                            data_bus->frames(),
+                            sizeof(render_audio[0]));
+  render_converter_->Push(data_bus.get());
+  while (render_converter_->Convert()) {
+    audio_processing_->AnalyzeReverseStream(render_converter_->audio_frame());
+  }
+}
+
+void WebRtcAudioProcessor::InitializeAudioProcessingModule(
+    const webrtc::MediaConstraintsInterface* constraints) {
+  const CommandLine& command_line = *CommandLine::ForCurrentProcess();
+  if (!command_line.HasSwitch(switches::kEnableWebRtcAudioProcessor))
+    return;
+
+  if (!constraints)
+    return;
+
+  bool enable_aec = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kEchoCancellation);
+  bool enable_experimental_aec = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kExperimentalEchoCancellation);
+  bool enable_ns = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kNoiseSuppression);
+  bool enable_high_pass_filter = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kHighpassFilter);
+  bool enable_typing_detection = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kTypingNoiseDetection);
+  // TODO(xians): How to start and stop AEC dump?
+  bool start_aec_dump = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kInternalAecDump);
+#if defined(IOS) || defined(ANDROID)
+  enable_typing_detection = false;
+  enable_experimental_aec = false;
+#endif
+
+  // Reset the audio processing to NULL if no audio processing component is
+  // enabled.
+  if (!enable_aec && !enable_experimental_aec && !enable_ns &&
+      !enable_high_pass_filter && !enable_typing_detection) {
+    return;
+  }
+
+  // Create and configure the audio processing if it does not exist.
+  if (!audio_processing_.get())
+    audio_processing_.reset(webrtc::AudioProcessing::Create(0));
+
+  // Enable the audio processing components.
+  if (enable_aec)
+    EnableEchoCancellation(audio_processing_.get());
+
+  if (enable_ns)
+    EnableNoiseSuppression(audio_processing_.get());
+
+  if (enable_high_pass_filter)
+    EnableHighPassFilter(audio_processing_.get());
+
+  if (enable_typing_detection)
+    EnableTypingDetection(audio_processing_.get());
+
+  if (enable_experimental_aec)
+    EnableExperimentalEchoCancellation(audio_processing_.get());
+
+  if (enable_aec && start_aec_dump)
+    StartAecDump(audio_processing_.get());
+
+  // Configure the audio format the audio processing is running on. This
+  // has to be done after all the needed components are enabled.
+  if (audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate))
+    NOTREACHED();
+  if (audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel,
+                                          kAudioProcessingNumberOfChannel))
+    NOTREACHED();
+}
+
+void WebRtcAudioProcessor::InitializeRenderConverterIfNeeded(
+    int sample_rate, int number_of_channels, int frames_per_buffer) {
+  // TODO, figure out if we need to handle the buffer size change.
+  if (render_converter_.get() &&
+      render_converter_->source_parameters().sample_rate() == sample_rate &&
+      render_converter_->source_parameters().channels() == number_of_channels) {
+    // Do nothing if the |render_converter_| is setup properly.
+    return;
+  }
+
+  media::AudioParameters source_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::GuessChannelLayout(number_of_channels), sample_rate, 16,
+      frames_per_buffer);
+  media::AudioParameters sink_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::CHANNEL_LAYOUT_MONO, kAudioProcessingSampleRate, 16,
+      kAudioProcessingSampleRate / 100);
+  render_converter_.reset(new WebRtcAudioConverter(source_params, sink_params));
+}
+
+void WebRtcAudioProcessor::StopAudioProcessing() {
+  if (!audio_processing_.get())
+    return;
+
+  // It is safe to stop the AEC dump even it is not started.
+  StopAecDump(audio_processing_.get());
+
+  audio_processing_.reset();
+}
+
+}  // namespace content